The introduction of low-cost high-density integrated circuits (VLSI) for PCM coding and decoding of speech signals has allowed direct digital transmission to and from a subscriber's telephone set. This enables both a higher quality of speech communication and a greater number of functions to be provided by the telephone line, such as data transmission and so on. Such transmission can be carried out, as in the traditional analog systems, by the simultaneous transmission of digital signals in both directions. The bandwidth occupied is maintained within acceptable values and the transmission line requires just two wires, so that existing subscriber lines, a coaxial cable, or an optical fiber can be used.
With this technique, the transmission directions are separated by an hybrid transformer which connects both the receiver and the transmitter to the line. An ideal hybrid balance and hence a complete separation between the two digital flows is however impossible to obtain.
In fact, the characteristics of existing telephone lines are different from the mean line characteristics for which the balancing network of the hybrid transformer is designed. The adjustment of the balancing network during its installation would be prohibitively expensive if it had to be carried out for each subscriber line.
Moreover the presence of possible signal reflections from impedance discontinuities along the line generates echo signals which cannot be eliminated by the hybrid balancing.
As a consequence, superposition of the two information flows occurs at the reception point, with a useful signal which can be less by various orders of magnitude than the disturbing echo signal (.perspectiveto.-40 dB). That makes a correct reception impossible, owing both to the degradation of the signal itself, and to the difficulty of extracting timing and synchronism information from the received data flow.
Hence the necessity of using the so-called echo cancellers, apt to eliminate the cross-talk between two digital flows at the receiving point. At the present two possible solutions are known for implementing the devices above. They are described in the paper entitled "Digital echo cancellation for baseband data transmission" N.A.M. Verhoeckx, et al IEEE Transactions on Acoustics, Speech, Signal Processing, Vol. ASSP-27-No.6, pages 768-781, December 1979.
The first solution requires several sampling operations per signalling period. An estimated echo-signal sample is subtracted from the obtained samples, so as to allow a faithful reconstruction of the useful signal by analog filtering. In a first phase, in which clock information is not available, cancellation takes place in an asynchronous way.
A phase locking between clocks of received and transmitted data is possible only when the reconstructed signal is considerably free of echo signal. The disadvantages of this technique are the high sampling rate needed to carry out numerous cancellations during the signalling period, and the impossibility of using automatic line equalizers. In fact the operation of these devices depends on received signal level, which in this case is altered by echo signal presence. Moreover systems of this kind cannot be fully integrated due to the presence of the imterpolator analog filter.
The second solution uses two samplings per signalling period. Once the two cancelling operations are over, a search is carried out for the optimal sampling phase by using suitable algorithms. Only after the canceller convergence is obtained, can the sampling phase be adjusted with respect to the clock of useful signal. Once the correction is effected, canceller convergence on the updated echo signal is awaited.
The disadvantage of this solution is that more cancellation cycles and phase adjustments are required to obtain optimal sampling instant, which entails a time loss while waiting for the attainment of the suitable conditions for correct communication. Moreover, two samplings per signalling period are required and hence the canceller operating rate is doubled. In this case too, insertion of an automatic equalizer is difficult.